Use the current default capture for the webaudio track to get the correct delay value

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 287 matching lines @@
   // Sets the |renderer_|, returns false if |renderer_| already exists.
   // Called on the main renderer thread.
   bool SetAudioRenderer(WebRtcAudioRenderer* renderer);
 
   // Adds/Removes the capturer to the ADM.
   // TODO(xians): Remove these two methods once the ADM does not need to pass
   // hardware information up to WebRtc.
   void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
   void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
 
+  // Gets the default capturer, which is the last capturer in |capturers_|.
+  scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
+
   // Gets paired device information of the capture device for the audio
   // renderer. This is used to pass on a session id, sample rate and buffer
   // size to a webrtc audio renderer (either local or remote), so that audio
   // will be rendered to a matching output device.
   // Returns true if the capture device has a paired output device, otherwise
   // false. Note that if there are more than one open capture device the
   // function will not be able to pick an appropriate device and return false.
   bool GetAuthorizedDeviceInfoForAudioRenderer(
       int* session_id, int* output_sample_rate, int* output_buffer_size);
 
@@ skipping 37 matching lines @@
   // Called on the AudioInputDevice worker thread.
   virtual void RenderData(uint8* audio_data,
                           int number_of_channels,
                           int number_of_frames,
                           int audio_delay_milliseconds) OVERRIDE;
 
   // Called on the main render thread.
   virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE;
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;
 
-  // Helper to get the default capturer, which is the last capturer in
-  // |capturers_|.
-  scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
-
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   int ref_count_;
 
   // List of captures which provides access to the native audio input layer
   // in the browser process.
   CapturerList capturers_;
 
   // Provides access to the audio renderer in the browser process.
@@ skipping 32 matching lines @@
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_